/* GStreamer * Copyright (C) 1999 Erik Walthinsen * Copyright (C) 2005 Edgard Lima * Copyright (C) 2005 Nokia Corporation * Copyright (C) 2007,2008 Axis Communications * * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public * License as published by the Free Software Foundation; either * version 2 of the License, or (at your option) any later version. * * This library is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Library General Public License for more details. * * You should have received a copy of the GNU Library General Public * License along with this library; if not, write to the * Free Software Foundation, Inc., 59 Temple Place - Suite 330, * Boston, MA 02111-1307, USA. */ #ifdef HAVE_CONFIG_H # include "config.h" #endif #include #include #include #include "gstrtpg726pay.h" GST_DEBUG_CATEGORY_STATIC (rtpg726pay_debug); #define GST_CAT_DEFAULT (rtpg726pay_debug) #define DEFAULT_FORCE_AAL2 TRUE enum { PROP_0, PROP_FORCE_AAL2, PROP_LAST }; static const GstElementDetails gst_rtp_g726_pay_details = GST_ELEMENT_DETAILS ("RTP G.726 payloader", "Codec/Payloader/Network", "Payload-encodes G.726 audio into a RTP packet", "Axis Communications "); static GstStaticPadTemplate gst_rtp_g726_pay_sink_template = GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-adpcm, " "channels = (int) 1, " "rate = (int) 8000, " "bitrate = (int) { 16000, 24000, 32000, 40000 }, " "layout = (string) \"g726\"") ); static GstStaticPadTemplate gst_rtp_g726_pay_src_template = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS ("application/x-rtp, " "media = (string) \"audio\", " "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", " "clock-rate = (int) 8000, " "encoding-name = (string) { \"G726-16\", \"G726-24\", \"G726-32\", \"G726-40\", " " \"AAL2-G726-16\", \"AAL2-G726-24\", \"AAL2-G726-32\", \"AAL2-G726-40\" } ") ); static void gst_rtp_g726_pay_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec); static void gst_rtp_g726_pay_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec); static gboolean gst_rtp_g726_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps); static GstFlowReturn gst_rtp_g726_pay_handle_buffer (GstBaseRTPPayload * payload, GstBuffer * buffer); GST_BOILERPLATE (GstRtpG726Pay, gst_rtp_g726_pay, GstBaseRTPAudioPayload, GST_TYPE_BASE_RTP_AUDIO_PAYLOAD); static void gst_rtp_g726_pay_base_init (gpointer klass) { GstElementClass *element_class = GST_ELEMENT_CLASS (klass); gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&gst_rtp_g726_pay_sink_template)); gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&gst_rtp_g726_pay_src_template)); gst_element_class_set_details (element_class, &gst_rtp_g726_pay_details); } static void gst_rtp_g726_pay_class_init (GstRtpG726PayClass * klass) { GObjectClass *gobject_class; GstBaseRTPPayloadClass *gstbasertppayload_class; gobject_class = (GObjectClass *) klass; gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass; gobject_class->set_property = gst_rtp_g726_pay_set_property; gobject_class->get_property = gst_rtp_g726_pay_get_property; g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_FORCE_AAL2, g_param_spec_boolean ("force-aal2", "Force AAL2", "Force AAL2 encoding for compatibility with bad depayloaders", DEFAULT_FORCE_AAL2, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); gstbasertppayload_class->set_caps = gst_rtp_g726_pay_setcaps; gstbasertppayload_class->handle_buffer = gst_rtp_g726_pay_handle_buffer; GST_DEBUG_CATEGORY_INIT (rtpg726pay_debug, "rtpg726pay", 0, "G.726 RTP Payloader"); } static void gst_rtp_g726_pay_init (GstRtpG726Pay * rtpg726pay, GstRtpG726PayClass * klass) { GstBaseRTPAudioPayload *basertpaudiopayload; basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (rtpg726pay); GST_BASE_RTP_PAYLOAD (rtpg726pay)->clock_rate = 8000; rtpg726pay->force_aal2 = DEFAULT_FORCE_AAL2; /* sample based codec */ gst_base_rtp_audio_payload_set_sample_based (basertpaudiopayload); } static gboolean gst_rtp_g726_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps) { gchar *encoding_name; GstStructure *structure; GstBaseRTPAudioPayload *basertpaudiopayload; GstRtpG726Pay *pay; GstCaps *peercaps; gboolean res; basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (payload); pay = GST_RTP_G726_PAY (payload); structure = gst_caps_get_structure (caps, 0); if (!gst_structure_get_int (structure, "bitrate", &pay->bitrate)) pay->bitrate = 32000; GST_DEBUG_OBJECT (payload, "using bitrate %d", pay->bitrate); pay->aal2 = FALSE; /* first see what we can do with the bitrate */ switch (pay->bitrate) { case 16000: encoding_name = g_strdup ("G726-16"); gst_base_rtp_audio_payload_set_samplebits_options (basertpaudiopayload, 2); break; case 24000: encoding_name = g_strdup ("G726-24"); gst_base_rtp_audio_payload_set_samplebits_options (basertpaudiopayload, 3); break; case 32000: encoding_name = g_strdup ("G726-32"); gst_base_rtp_audio_payload_set_samplebits_options (basertpaudiopayload, 4); break; case 40000: encoding_name = g_strdup ("G726-40"); gst_base_rtp_audio_payload_set_samplebits_options (basertpaudiopayload, 5); break; default: goto invalid_bitrate; } GST_DEBUG_OBJECT (payload, "selected base encoding %s", encoding_name); /* now see if we need to produce AAL2 or not */ peercaps = gst_pad_peer_get_caps (payload->srcpad); if (peercaps) { GstCaps *filter, *intersect; gchar *capsstr; GST_DEBUG_OBJECT (payload, "have peercaps %" GST_PTR_FORMAT, peercaps); capsstr = g_strdup_printf ("application/x-rtp, " "media = (string) \"audio\", " "clock-rate = (int) 8000, " "encoding-name = (string) %s; " "application/x-rtp, " "media = (string) \"audio\", " "clock-rate = (int) 8000, " "encoding-name = (string) AAL2-%s", encoding_name, encoding_name); filter = gst_caps_from_string (capsstr); g_free (capsstr); /* intersect to filter */ intersect = gst_caps_intersect (peercaps, filter); gst_caps_unref (peercaps); GST_DEBUG_OBJECT (payload, "intersected to %" GST_PTR_FORMAT, intersect); if (!intersect) goto no_format; if (gst_caps_is_empty (intersect)) { gst_caps_unref (intersect); goto no_format; } structure = gst_caps_get_structure (intersect, 0); /* now see what encoding name we settled on, we need to dup because the * string goes away when we unref the intersection below. */ encoding_name = g_strdup (gst_structure_get_string (structure, "encoding-name")); /* if we managed to negotiate to AAL2, we definatly are going to do AAL2 * encoding. Else we only encode AAL2 when explicitly set by the * property. */ if (g_str_has_prefix (encoding_name, "AAL2-")) pay->aal2 = TRUE; else pay->aal2 = pay->force_aal2; GST_DEBUG_OBJECT (payload, "final encoding %s, AAL2 %d", encoding_name, pay->aal2); gst_caps_unref (intersect); } else { /* downstream can do anything but we prefer the better supported non-AAL2 */ pay->aal2 = pay->force_aal2; GST_DEBUG_OBJECT (payload, "no peer caps, AAL2 %d", pay->aal2); } gst_basertppayload_set_options (payload, "audio", TRUE, encoding_name, 8000); res = gst_basertppayload_set_outcaps (payload, NULL); g_free (encoding_name); return res; /* ERRORS */ invalid_bitrate: { GST_ERROR_OBJECT (payload, "invalid bitrate %d specified", pay->bitrate); return FALSE; } no_format: { GST_ERROR_OBJECT (payload, "could not negotiate format"); return FALSE; } } static GstFlowReturn gst_rtp_g726_pay_handle_buffer (GstBaseRTPPayload * payload, GstBuffer * buffer) { GstFlowReturn res; GstRtpG726Pay *pay; pay = GST_RTP_G726_PAY (payload); if (!pay->aal2) { guint8 *data, tmp; guint len; /* for non AAL2, we need to reshuffle the bytes, we can do this in-place * when the buffer is writable. */ buffer = gst_buffer_make_writable (buffer); data = GST_BUFFER_DATA (buffer); len = GST_BUFFER_SIZE (buffer); GST_LOG_OBJECT (pay, "packing %u bytes of data", len); /* we need to reshuffle the bytes, output is of the form: * A B C D .. with the number of bits depending on the bitrate. */ switch (pay->bitrate) { case 16000: { /* 0 * 0 1 2 3 4 5 6 7 * +-+-+-+-+-+-+-+-+- * |D D|C C|B B|A A| ... * |0 1|0 1|0 1|0 1| * +-+-+-+-+-+-+-+-+- */ while (len > 0) { tmp = *data; *data++ = ((tmp & 0xc0) >> 6) | ((tmp & 0x30) >> 2) | ((tmp & 0x0c) << 2) | ((tmp & 0x03) << 6); len--; } break; } case 24000: { /* 0 1 2 * 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+- * |C C|B B B|A A A|F|E E E|D D D|C|H H H|G G G|F F| ... * |1 2|0 1 2|0 1 2|2|0 1 2|0 1 2|0|0 1 2|0 1 2|0 1| * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+- */ while (len > 2) { tmp = *data; *data++ = ((tmp & 0xc0) >> 6) | ((tmp & 0x38) >> 1) | ((tmp & 0x07) << 5); tmp = *data; *data++ = ((tmp & 0x80) >> 7) | ((tmp & 0x70) >> 3) | ((tmp & 0x0e) << 4) | ((tmp & 0x01) << 7); tmp = *data; *data++ = ((tmp & 0xe0) >> 5) | ((tmp & 0x1c) >> 2) | ((tmp & 0x03) << 6); len -= 3; } break; } case 32000: { /* 0 1 * 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+- * |B B B B|A A A A|D D D D|C C C C| ... * |0 1 2 3|0 1 2 3|0 1 2 3|0 1 2 3| * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+- */ while (len > 0) { tmp = *data; *data++ = ((tmp & 0xf0) >> 4) | ((tmp & 0x0f) << 4); len--; } break; } case 40000: { /* 0 1 2 3 4 * 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+- * |B B B|A A A A A|D|C C C C C|B B|E E E E|D D D D|G G|F F F F F|E|H H H H H|G G G| * |2 3 4|0 1 2 3 4|4|0 1 2 3 4|0 1|1 2 3 4|0 1 2 3|3 4|0 1 2 3 4|0|0 1 2 3 4|0 1 2| * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+- */ while (len > 4) { tmp = *data; *data++ = ((tmp & 0xe0) >> 5) | ((tmp & 0x1f) << 3); tmp = *data; *data++ = ((tmp & 0x80) >> 7) | ((tmp & 0x7c) >> 2) | ((tmp & 0x03) << 6); tmp = *data; *data++ = ((tmp & 0xf0) >> 4) | ((tmp & 0x0f) << 4); tmp = *data; *data++ = ((tmp & 0xc0) >> 6) | ((tmp & 0x3e) << 2) | ((tmp & 0x01) << 7); tmp = *data; *data++ = ((tmp & 0xf8) >> 3) | ((tmp & 0x07) << 5); len -= 5; } break; } } } res = GST_BASE_RTP_PAYLOAD_CLASS (parent_class)->handle_buffer (payload, buffer); return res; } static void gst_rtp_g726_pay_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec) { GstRtpG726Pay *rtpg726pay; rtpg726pay = GST_RTP_G726_PAY (object); switch (prop_id) { case PROP_FORCE_AAL2: rtpg726pay->force_aal2 = g_value_get_boolean (value); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static void gst_rtp_g726_pay_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec) { GstRtpG726Pay *rtpg726pay; rtpg726pay = GST_RTP_G726_PAY (object); switch (prop_id) { case PROP_FORCE_AAL2: g_value_set_boolean (value, rtpg726pay->force_aal2); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } gboolean gst_rtp_g726_pay_plugin_init (GstPlugin * plugin) { return gst_element_register (plugin, "rtpg726pay", GST_RANK_NONE, GST_TYPE_RTP_G726_PAY); }